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Re: [webrtc-stats] Stats to keep track of sync between audio and video

From: henbos via GitHub <sysbot+gh@w3.org>
Date: Thu, 11 May 2017 08:32:40 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-300720813-1494491558-sysbot+gh@w3.org>
See https://github.com/w3c/webrtc-stats/pull/187#pullrequestreview-37254569 for excluding estimatedCaptureTimestamp:

> captureTimestamp removed after offline discussion: The RTCP SR packet containing the NTP timestamp doesn't give us a RTT estimate because the local and remote clocks aren't in sync.
> If the alternative is to use RTT estimates from STUN pings then we're ignoring delays due to encoding and packetization and using an RTT from other packets than the media packets. If this is acceptable in the estimatedCaptureTimestamp this can already be calculated by taking track.estimatedPlayoutTimestamp + iceCandidatePair.currentRoundTripTime / 2 and we don't need a new stat for it that could be slightly misleading due to the difference between media packets and stun pings.

GitHub Notification of comment by henbos
Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/158#issuecomment-300720813 using your GitHub account
Received on Thursday, 11 May 2017 08:32:46 UTC

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